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Pass in the Q parameter for setting the filter parameters

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Also better handle the peaking filter gain.
This commit is contained in:
Chris Robinson 2015-11-01 04:43:55 -08:00
parent f094d94608
commit c57f571920
9 changed files with 116 additions and 64 deletions
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40
Alc/ALu.c
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@ -667,18 +667,20 @@ ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const A
{
ALfloat hfscale = ALSource->Direct.HFReference / Frequency;
ALfloat lfscale = ALSource->Direct.LFReference / Frequency;
DryGainHF = maxf(DryGainHF, 0.0001f);
DryGainLF = maxf(DryGainLF, 0.0001f);
for(c = 0;c < num_channels;c++)
{
voice->Direct.Filters[c].ActiveType = AF_None;
if(DryGainHF != 1.0f) voice->Direct.Filters[c].ActiveType |= AF_LowPass;
if(DryGainLF != 1.0f) voice->Direct.Filters[c].ActiveType |= AF_HighPass;
ALfilterState_setParams(
&voice->Direct.Filters[c].LowPass, ALfilterType_HighShelf, DryGainHF,
hfscale, 0.0f
&voice->Direct.Filters[c].LowPass, ALfilterType_HighShelf,
DryGainHF, hfscale, calc_rcpQ_from_slope(DryGainHF, 0.75f)
);
ALfilterState_setParams(
&voice->Direct.Filters[c].HighPass, ALfilterType_LowShelf, DryGainLF,
lfscale, 0.0f
&voice->Direct.Filters[c].HighPass, ALfilterType_LowShelf,
DryGainLF, lfscale, calc_rcpQ_from_slope(DryGainLF, 0.75f)
);
}
}
@ -686,18 +688,20 @@ ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const A
{
ALfloat hfscale = ALSource->Send[i].HFReference / Frequency;
ALfloat lfscale = ALSource->Send[i].LFReference / Frequency;
WetGainHF[i] = maxf(WetGainHF[i], 0.0001f);
WetGainLF[i] = maxf(WetGainLF[i], 0.0001f);
for(c = 0;c < num_channels;c++)
{
voice->Send[i].Filters[c].ActiveType = AF_None;
if(WetGainHF[i] != 1.0f) voice->Send[i].Filters[c].ActiveType |= AF_LowPass;
if(WetGainLF[i] != 1.0f) voice->Send[i].Filters[c].ActiveType |= AF_HighPass;
ALfilterState_setParams(
&voice->Send[i].Filters[c].LowPass, ALfilterType_HighShelf, WetGainHF[i],
hfscale, 0.0f
&voice->Send[i].Filters[c].LowPass, ALfilterType_HighShelf,
WetGainHF[i], hfscale, calc_rcpQ_from_slope(WetGainHF[i], 0.75f)
);
ALfilterState_setParams(
&voice->Send[i].Filters[c].HighPass, ALfilterType_LowShelf, WetGainLF[i],
lfscale, 0.0f
&voice->Send[i].Filters[c].HighPass, ALfilterType_LowShelf,
WetGainLF[i], lfscale, calc_rcpQ_from_slope(WetGainLF[i], 0.75f)
);
}
}
@ -1139,32 +1143,36 @@ ALvoid CalcSourceParams(ALvoice *voice, const ALsource *ALSource, const ALCconte
{
ALfloat hfscale = ALSource->Direct.HFReference / Frequency;
ALfloat lfscale = ALSource->Direct.LFReference / Frequency;
DryGainHF = maxf(DryGainHF, 0.0001f);
DryGainLF = maxf(DryGainLF, 0.0001f);
voice->Direct.Filters[0].ActiveType = AF_None;
if(DryGainHF != 1.0f) voice->Direct.Filters[0].ActiveType |= AF_LowPass;
if(DryGainLF != 1.0f) voice->Direct.Filters[0].ActiveType |= AF_HighPass;
ALfilterState_setParams(
&voice->Direct.Filters[0].LowPass, ALfilterType_HighShelf, DryGainHF,
hfscale, 0.0f
&voice->Direct.Filters[0].LowPass, ALfilterType_HighShelf,
DryGainHF, hfscale, calc_rcpQ_from_slope(DryGainHF, 0.75f)
);
ALfilterState_setParams(
&voice->Direct.Filters[0].HighPass, ALfilterType_LowShelf, DryGainLF,
lfscale, 0.0f
&voice->Direct.Filters[0].HighPass, ALfilterType_LowShelf,
DryGainLF, lfscale, calc_rcpQ_from_slope(DryGainLF, 0.75f)
);
}
for(i = 0;i < NumSends;i++)
{
ALfloat hfscale = ALSource->Send[i].HFReference / Frequency;
ALfloat lfscale = ALSource->Send[i].LFReference / Frequency;
WetGainHF[i] = maxf(WetGainHF[i], 0.0001f);
WetGainLF[i] = maxf(WetGainLF[i], 0.0001f);
voice->Send[i].Filters[0].ActiveType = AF_None;
if(WetGainHF[i] != 1.0f) voice->Send[i].Filters[0].ActiveType |= AF_LowPass;
if(WetGainLF[i] != 1.0f) voice->Send[i].Filters[0].ActiveType |= AF_HighPass;
ALfilterState_setParams(
&voice->Send[i].Filters[0].LowPass, ALfilterType_HighShelf, WetGainHF[i],
hfscale, 0.0f
&voice->Send[i].Filters[0].LowPass, ALfilterType_HighShelf,
WetGainHF[i], hfscale, calc_rcpQ_from_slope(WetGainHF[i], 0.75f)
);
ALfilterState_setParams(
&voice->Send[i].Filters[0].HighPass, ALfilterType_LowShelf, WetGainLF[i],
lfscale, 0.0f
&voice->Send[i].Filters[0].HighPass, ALfilterType_LowShelf,
WetGainLF[i], lfscale, calc_rcpQ_from_slope(WetGainLF[i], 0.75f)
);
}
}

6
Alc/effects/distortion.c
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@ -72,14 +72,16 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, ALCdevice *Devi
/* Bandwidth value is constant in octaves */
bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
ALfilterState_setParams(&state->lowpass, ALfilterType_LowPass, 1.0f,
cutoff / (frequency*4.0f), bandwidth);
cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
);
/* Bandpass filter */
cutoff = Slot->EffectProps.Distortion.EQCenter;
/* Convert bandwidth in Hz to octaves */
bandwidth = Slot->EffectProps.Distortion.EQBandwidth / (cutoff * 0.67f);
ALfilterState_setParams(&state->bandpass, ALfilterType_BandPass, 1.0f,
cutoff / (frequency*4.0f), bandwidth);
cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
);
ComputeAmbientGains(Device, Slot->Gain, state->Gain);
}

10
Alc/effects/echo.c
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@ -85,8 +85,7 @@ static ALvoid ALechoState_update(ALechoState *state, ALCdevice *Device, const AL
{
ALfloat pandir[3] = { 0.0f, 0.0f, 0.0f };
ALuint frequency = Device->Frequency;
ALfloat gain = Slot->Gain;
ALfloat lrpan;
ALfloat gain, lrpan;
state->Tap[0].delay = fastf2u(Slot->EffectProps.Echo.Delay * frequency) + 1;
state->Tap[1].delay = fastf2u(Slot->EffectProps.Echo.LRDelay * frequency);
@ -96,9 +95,12 @@ static ALvoid ALechoState_update(ALechoState *state, ALCdevice *Device, const AL
state->FeedGain = Slot->EffectProps.Echo.Feedback;
gain = minf(1.0f - Slot->EffectProps.Echo.Damping, 0.01f);
ALfilterState_setParams(&state->Filter, ALfilterType_HighShelf,
1.0f - Slot->EffectProps.Echo.Damping,
LOWPASSFREQREF/frequency, 0.0f);
gain, LOWPASSFREQREF/frequency,
calc_rcpQ_from_slope(gain, 0.75f));
gain = Slot->Gain;
/* First tap panning */
pandir[0] = -lrpan;

34
Alc/effects/equalizer.c
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@ -93,29 +93,37 @@ static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state),
static ALvoid ALequalizerState_update(ALequalizerState *state, ALCdevice *device, const ALeffectslot *slot)
{
ALfloat frequency = (ALfloat)device->Frequency;
ALfloat gain, freq_mult;
ComputeAmbientGains(device, slot->Gain, state->Gain);
/* Calculate coefficients for the each type of filter */
/* Calculate coefficients for the each type of filter. Note that the shelf
* filters' gain is for the reference frequency, which is the centerpoint
* of the transition band.
*/
gain = sqrtf(slot->EffectProps.Equalizer.LowGain);
freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency;
ALfilterState_setParams(&state->filter[0], ALfilterType_LowShelf,
sqrtf(slot->EffectProps.Equalizer.LowGain),
slot->EffectProps.Equalizer.LowCutoff/frequency,
0.0f);
gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
);
gain = slot->EffectProps.Equalizer.Mid1Gain;
freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency;
ALfilterState_setParams(&state->filter[1], ALfilterType_Peaking,
sqrtf(slot->EffectProps.Equalizer.Mid1Gain),
slot->EffectProps.Equalizer.Mid1Center/frequency,
slot->EffectProps.Equalizer.Mid1Width);
gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid1Width)
);
gain = slot->EffectProps.Equalizer.Mid2Gain;
freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency;
ALfilterState_setParams(&state->filter[2], ALfilterType_Peaking,
sqrtf(slot->EffectProps.Equalizer.Mid2Gain),
slot->EffectProps.Equalizer.Mid2Center/frequency,
slot->EffectProps.Equalizer.Mid2Width);
gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid2Width)
);
gain = sqrtf(slot->EffectProps.Equalizer.HighGain);
freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency;
ALfilterState_setParams(&state->filter[3], ALfilterType_HighShelf,
sqrtf(slot->EffectProps.Equalizer.HighGain),
slot->EffectProps.Equalizer.HighCutoff/frequency,
0.0f);
gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
);
}
static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)

7
Alc/effects/reverb.c
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@ -1112,6 +1112,7 @@ static ALvoid ALreverbState_update(ALreverbState *State, ALCdevice *Device, cons
const ALeffectProps *props = &Slot->EffectProps;
ALuint frequency = Device->Frequency;
ALfloat lfscale, hfscale, hfRatio;
ALfloat gainlf, gainhf;
ALfloat cw, x, y;
if(Slot->EffectType == AL_EFFECT_EAXREVERB && !EmulateEAXReverb)
@ -1121,11 +1122,13 @@ static ALvoid ALreverbState_update(ALreverbState *State, ALCdevice *Device, cons
// Calculate the master filters
hfscale = props->Reverb.HFReference / frequency;
gainhf = maxf(props->Reverb.GainHF, 0.0001f);
ALfilterState_setParams(&State->LpFilter, ALfilterType_HighShelf,
props->Reverb.GainHF, hfscale, 0.0f);
gainhf, hfscale, calc_rcpQ_from_slope(gainhf, 0.75f));
lfscale = props->Reverb.LFReference / frequency;
gainlf = maxf(props->Reverb.GainLF, 0.0001f);
ALfilterState_setParams(&State->HpFilter, ALfilterType_LowShelf,
props->Reverb.GainLF, lfscale, 0.0f);
gainlf, lfscale, calc_rcpQ_from_slope(gainlf, 0.75f));
// Update the modulator line.
UpdateModulator(props->Reverb.ModulationTime, props->Reverb.ModulationDepth,

43
OpenAL32/Include/alFilter.h
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@ -3,6 +3,8 @@
#include "alMain.h"
#include "math_defs.h"
#ifdef __cplusplus
extern "C" {
#endif
@ -11,23 +13,29 @@ extern "C" {
#define HIGHPASSFREQREF (250.0f)
/* Filters implementation is based on the "Cookbook formulae for audio *
* EQ biquad filter coefficients" by Robert Bristow-Johnson *
* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */
/* Filters implementation is based on the "Cookbook formulae for audio
* EQ biquad filter coefficients" by Robert Bristow-Johnson
* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
*/
/* Implementation note: For the shelf filters, the specified gain is for the
* reference frequency, which is the centerpoint of the transition band. This
* better matches EFX filter design. To set the gain for the shelf itself, use
* the square root of the desired linear gain (or halve the dB gain).
*/
typedef enum ALfilterType {
/** EFX-style low-pass filter, specifying a gain and reference frequency. */
ALfilterType_HighShelf,
/** EFX-style high-pass filter, specifying a gain and reference frequency. */
ALfilterType_LowShelf,
/** Peaking filter, specifying a gain, reference frequency, and bandwidth. */
/** Peaking filter, specifying a gain and reference frequency. */
ALfilterType_Peaking,
/** Low-pass cut-off filter, specifying a cut-off frequency and bandwidth. */
/** Low-pass cut-off filter, specifying a cut-off frequency. */
ALfilterType_LowPass,
/** High-pass cut-off filter, specifying a cut-off frequency and bandwidth. */
/** High-pass cut-off filter, specifying a cut-off frequency. */
ALfilterType_HighPass,
/** Band-pass filter, specifying a center frequency and bandwidth. */
/** Band-pass filter, specifying a center frequency. */
ALfilterType_BandPass,
} ALfilterType;
@ -41,8 +49,27 @@ typedef struct ALfilterState {
} ALfilterState;
#define ALfilterState_process(a, ...) ((a)->process((a), __VA_ARGS__))
/* Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the
* reference gain and shelf slope parameter.
* 0 < gain
* 0 < slope <= 1
*/
inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope)
{
return sqrtf((gain + 1.0f/gain)*(1.0f/slope - 1.0f) + 2.0f);
}
/* Calculates the rcpQ (i.e. 1/Q) coefficient for filters, using the frequency
* multiple (i.e. ref_freq / sampling_freq) and bandwidth.
* 0 < freq_mult < 0.5.
*/
inline ALfloat calc_rcpQ_from_bandwidth(ALfloat freq_mult, ALfloat bandwidth)
{
ALfloat w0 = F_TAU * freq_mult;
return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0));
}
void ALfilterState_clear(ALfilterState *filter);
void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat bandwidth);
void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat rcpQ);
inline ALfloat ALfilterState_processSingle(ALfilterState *filter, ALfloat sample)
{

13
OpenAL32/Include/alu.h
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@ -16,18 +16,7 @@
#include "hrtf.h"
#include "align.h"
#define F_PI (3.14159265358979323846f)
#define F_PI_2 (1.57079632679489661923f)
#define F_TAU (6.28318530717958647692f)
#ifndef FLT_EPSILON
#define FLT_EPSILON (1.19209290e-07f)
#endif
#define DEG2RAD(x) ((ALfloat)(x) * (F_PI/180.0f))
#define RAD2DEG(x) ((ALfloat)(x) * (180.0f/F_PI))
#include "math_defs.h"
#define MAX_PITCH (255)

12
OpenAL32/alFilter.c
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@ -32,6 +32,8 @@
extern inline struct ALfilter *LookupFilter(ALCdevice *device, ALuint id);
extern inline struct ALfilter *RemoveFilter(ALCdevice *device, ALuint id);
extern inline ALfloat ALfilterState_processSingle(ALfilterState *filter, ALfloat sample);
extern inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope);
extern inline ALfloat calc_rcpQ_from_bandwidth(ALfloat freq_mult, ALfloat bandwidth);
static void InitFilterParams(ALfilter *filter, ALenum type);
@ -336,7 +338,7 @@ void ALfilterState_clear(ALfilterState *filter)
filter->y[1] = 0.0f;
}
void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat bandwidth)
void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat rcpQ)
{
ALfloat alpha, sqrtgain_alpha_2;
ALfloat w0, sin_w0, cos_w0;
@ -347,12 +349,12 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
w0 = F_TAU * freq_mult;
sin_w0 = sinf(w0);
cos_w0 = cosf(w0);
alpha = sin_w0/2.0f * rcpQ;
/* Calculate filter coefficients depending on filter type */
switch(type)
{
case ALfilterType_HighShelf:
alpha = sin_w0/2.0f*sqrtf((gain + 1.0f/gain)*(1.0f/0.75f - 1.0f) + 2.0f);
sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha;
filter->b[0] = gain*((gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
filter->b[1] = -2.0f*gain*((gain-1.0f) + (gain+1.0f)*cos_w0 );
@ -362,7 +364,6 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
filter->a[2] = (gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
break;
case ALfilterType_LowShelf:
alpha = sin_w0/2.0f*sqrtf((gain + 1.0f/gain)*(1.0f/0.75f - 1.0f) + 2.0f);
sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha;
filter->b[0] = gain*((gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
filter->b[1] = 2.0f*gain*((gain-1.0f) - (gain+1.0f)*cos_w0 );
@ -372,7 +373,7 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
filter->a[2] = (gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
break;
case ALfilterType_Peaking:
alpha = sin_w0 * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sin_w0);
gain = sqrtf(gain);
filter->b[0] = 1.0f + alpha * gain;
filter->b[1] = -2.0f * cos_w0;
filter->b[2] = 1.0f - alpha * gain;
@ -382,7 +383,6 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
break;
case ALfilterType_LowPass:
alpha = sin_w0 * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sin_w0);
filter->b[0] = (1.0f - cos_w0) / 2.0f;
filter->b[1] = 1.0f - cos_w0;
filter->b[2] = (1.0f - cos_w0) / 2.0f;
@ -391,7 +391,6 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
filter->a[2] = 1.0f - alpha;
break;
case ALfilterType_HighPass:
alpha = sin_w0 * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sin_w0);
filter->b[0] = (1.0f + cos_w0) / 2.0f;
filter->b[1] = -(1.0f + cos_w0);
filter->b[2] = (1.0f + cos_w0) / 2.0f;
@ -400,7 +399,6 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
filter->a[2] = 1.0f - alpha;
break;
case ALfilterType_BandPass:
alpha = sin_w0 * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sin_w0);
filter->b[0] = alpha;
filter->b[1] = 0;
filter->b[2] = -alpha;

15
include/math_defs.h Normal file
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@ -0,0 +1,15 @@
#ifndef AL_MATH_DEFS_H
#define AL_MATH_DEFS_H
#define F_PI (3.14159265358979323846f)
#define F_PI_2 (1.57079632679489661923f)
#define F_TAU (6.28318530717958647692f)
#ifndef FLT_EPSILON
#define FLT_EPSILON (1.19209290e-07f)
#endif
#define DEG2RAD(x) ((ALfloat)(x) * (F_PI/180.0f))
#define RAD2DEG(x) ((ALfloat)(x) * (180.0f/F_PI))
#endif /* AL_MATH_DEFS_H */